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Three factors have been often shown to significantly affect the reliability of real-time transmission in wireless local area networks-transmission rate, power, and packet size. We analyze these factors and determine the optimal combination of factors subject to time constraints that yields the most reliable transmission of real-time data. We propose a cross-layer framework to jointly design the routing and MAC protocol combined with our optimization approach. The approach under a non-real-time routing protocol that produces a path metric is compared with a real-time routing protocol. Additionally, the real-time routing protocol enhances a guaranteed rate with our approach. Our experiments reveal that real-time performance in terms of miss ratio and throughput is significantly increased in lossy link and heavy traffic environments. Miss ratio and average throughput are improved by up to 30% over a state-of-the art routing protocol and 35% over a MAC protocol, respectively. [ABSTRACT FROM AUTHOR]

Medium access control overhead is the primary reason for low throughput in wireless networks. Performing blind contentions, contentions without any information of other contenders, and exchanging control message are time-consuming control operations. In this study, we propose a new MAC protocol called distributed sequential access MAC (DSA-MAC) which provides the transmission order without any explicit control operations. It may induce very light control overhead; therefore, compared to existing wireless MAC protocols, DSA-MAC can remarkably enhance network throughput. [ABSTRACT FROM AUTHOR]

For extremely sensitive applications, it may be advantageous for users to transmit certain types of data covertly over the network. This provides an additional layer of security to that provided by the different layers of the protocol stack. In this paper we present a covert side channel that uses the 802.11 MAC rate switching protocol. The covert channel provides a general method to hide communications within currently deployed 802.11 LANs. The technique uses a one-time password (OTP) algorithm to ensure high-entropy randomness of the covert messages. We investigate how the covert side channel affects network throughput under various rate-switching conditions with UDP-based and TCP-based application traffic. We also investigate the covertness of the covert side channel using standardized entropy. The theoretical analysis shows that the maximum covert channel bandwidth is 60 bps. The simulation results show that the impact on network throughput is minimal and increases slightly as the covert channel bandwidth increases. We further show that the channel has 100% accuracy with minimal impact on rate switching entropy for scenarios where rate switching normally occurs. Finally, we present two applications for the covert channel: covert authentication and covert WiFi botnets. Copyright © 2010 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

Amongst link technologies, InfiniBand has gained wide acceptance in the framework of High Performance Computing (HPC), due to its high bandwidth and in particular to its low latency. Since InfiniBand is very flexible, supporting several kinds of messages, it is suitable, in principle, not only for HPC, but also for the data acquisition systems of High Energy Physics (HEP) Experiments. [ABSTRACT FROM PUBLISHER]

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Abstract: In this paper, we present a two-stage forward error correction (FEC) scheme with an enhanced link-layer protocol especially for multimedia data transmission over wireless LANs. At the application layer, packet-level FEC (stage-one) is added across packets to correct packet losses due to congestion and route disruption. Bit-level FEC (stage-two) is then added to both application packets and stage-one FEC packets to recover bit errors from the link layer. Then at the link layer, header-CRC/FEC is used to enhance protection and to cooperate with the two-stage FEC scheme. The proposed scheme thus provides joint protection across the protocol stack. We explore both its bandwidth efficiency and video performance for the highly efficient and scalable MC-EZBC video codec using the network simulator ns-2. Our results show that the proposed scheme can effectively increase application-layer throughput, reduce both end-to-end transmission delay and application bandwidth fluctuation, and significantly improve video performance. [Copyright &y& Elsevier]

The advent of IP/MPLS based networks allows providers to implement Integrated Services Packet Network (ISPN) concepts for IPTV applications. IPTV systems require a high level of Quality of Service (QoS) in order to win customers which are subscribed with cable companies. Packet loss probability is one of the primary QoS parameters whose value affects the user experience and which provides a quantitative measure for customers' perception of the QoS factor in practical networks. Therefore, the online accurate measurement and estimation of the packet loss probability is a key issue to be addressed. Furthermore, the packet loss probability provides the feedback information which can be used in the network based control system to maintain it at a prescribed and negotiated value. In this paper, a Reactive Estimator (RE) of the packet loss probability is constructed. As all parameters related to the network traffic, the packet loss is a non-linear and non-Gaussian stochastic process. The proposed RE has the capability to adapt to variable stochastic contexts by employing one dynamic item based on the feedback of the real-time loss ratio measurement. A series of experiments are devised on a live network to evaluate the performance of the estimator under multiple traffic arrival models and within various buffer sizes. The numeric results show that the practical estimator is accurate enough to approximate the packet loss probability, such that it can be further used as a feedback parameter in a closed control loop which can keep the packet loss probability close to a prescribed value. [ABSTRACT FROM AUTHOR]

Internet Protocol TV (IPTV), an emerging internet application, provides more flexibility and interactivity for users because of its embedded return channel. Wireless 802.11 networks, used as the last hop of IPTV networks, would add another dimension of mobility, which is of great value for hot spot deployment and is increasingly required for in-home content distribution. However, a wireless channel may suffer from multi-path fading and interference, which may cause random and burst packet loss and impact users' quality of experience of an IPTV program. We report the design, implementation and evaluation of Merged Hybrid Adaptive FEC and ARQ (MHARQ) system for IPTV multicast over wireless LANs. In MHARQ, an IPTV cache server is introduced to improve the reliability for the last hop wireless distribution. MHARQ combines the advantages of receiver-driven staggered FEC and hybrid ARQ schemes to compensate the large dynamic range of WLAN channels and to achieve high reliability, scalability and wireless bandwidth efficiency for IPTV multicast. We also address backward compatible FEC encoding and decoding, audio and video synchronization issues in practical system implementation. Using the ORBIT radio grid test bed, we have investigated the performance of the proposed MHARQ system with various numbers of users per AP and different numbers of APs per IPTV cache server. It is demonstrated via real system implementation on ORBIT that MHARQ improves wireless bandwidth efficiency and scalability for reliable IPTV multicast, compared with existing reliable multicast schemes. [ABSTRACT FROM AUTHOR]

Abstract: In the past three decades, tremendous Ethernet-related research has been done, which has led to today''s ubiquitous Ethernet technology. On the other hand, with the emergence of new network needs, a new protocol, the IEEE 1394 standard serial bus (or Firewire) was introduced. Firewire is suitable for high-quality audio/video applications which do not perform well in the best-effort-based Ethernet technology. However, since Firewire is a serial bus, it has harsh cable length limitations as compared to Ethernet capabilities. In this paper, we present a novel on-chip system that receives Firewire video and transmits it in multicast mode using Ethernet protocol. A major advantage of this novel system is to utilize the existing Ethernet infrastructure to extend the range of Firewire video streaming to reach remote nodes and make it even accessible to nodes with a single Ethernet interface. This will have tremendous impact on Firewire applications such as deploying Firewire cameras in big-scale security-sensitive buildings or industrial facilities with image-based remote quality control. This novel chip utilizes the concept of Ethernet multicasting transmission mode for video streaming. The proposed chip design converts the IEEE 1394 isochronous traffic to the Ethernet multicast frame format via two off-chip asynchronous write and read buffers. The goal of this research is to design an On Chip Novel Video Streaming System that avoids performance bottlenecks in the software protocol conversion of these two important network protocols. The author decided to study these two networks because of their broad use and cable power provisioning capabilities. The novel system design is implemented using a customized field programmable gate array (FPGA), which enables the integration of various system components on one chip. The designed prototype is studied using both network monitoring tools and analytical techniques, to verify its function and compare it with the existing approaches. Performance measures show that the On Chip Novel Video Streaming System consumes less than 21mW of power for 100Mbps and 82mW of power for 1Gbps, and utilizes 57% of a Xilinx Spartan 2-100E-6FT256 FPGA resources. Hence, it is possible to incorporate further extensions. Experimental results show that 88% of the network utilization can be achieved, due to the use of the customized, FPGA-based design of bi-network traffic conversion. [Copyright &y& Elsevier]

Wireless Sensor Network (WSN) is one of the major research areas in computer network field today. The function of WSN in this paper is to provide sensing services in an un-attended harsh environment. Sensed data need to be delivered to the base station and to cope with the network unreliability problem. Few routing protocol takes into consideration of this problem. It is a great challenge of the hierarchical routing protocol to provide network survivability through redundancy features. In this paper, a short literature review of the existing routing protocol is carried out. Then a novel hierarchical routing protocol, which addresses network survivability and redundancy issues, is introduced. Initial analysis shows promising results of the proposed protocol comparing with LEACH, which is a well known protocol as benchmark. Finally, conclusion was drawn based on the research and future direction for further research is identified. [ABSTRACT FROM AUTHOR]

A common communications convergence scenario which is being adopted in personal communications relates to the combination of wireless and cellular networks by the use of multimode terminals. Since most of the wireless networks were initially dimensioned only for data communications, this paper shows how voice over wireless LAN dimensioning could be addressed under the optimal network throughput and the perspective of voice quality, using a simple approach. The maximum number of simultaneous users resulting from throughput is limited by the collisions taking place in the shared medium with the statistical contention protocol. The voice quality is conditioned by the delay and the packet loss in the contention protocol. Both approaches are analyzed within the scope of the voice codecs commonly used in voice over wireless LANs, to conclude that voice dimensioning based on network throughput and voice quality show complementary results. Additionally the use of low rate codecs in voice over wireless LANs is advantageous for the network performance point of view but may produce poor voice quality results. Mid range codecs like G729 could represent a trade-off for quality throughput. For these reasons, voice quality and wireless network throughput have to be taken into account in the network admission control, design and deployment to ensure a satisfactory user experience. The impact of handoff interval of wireless convergent networks on the conversation quality need also be assessed for a proper network design. [ABSTRACT FROM AUTHOR]

As Ethernet traffic increases rapidly in carrier networks, service providers have an increasing need for Ethernet infrastructure networking in order to scale their Ethernet services. Ethernet infrastructure transports Ethernet traffic over distance, protects the traffic from link and nodal failures, and aggregates Ethernet connections from a large number of low-speed ports onto a much smaller number of high-speed ports. An Ethernet infrastructure allows service providers to inexpensively interconnect end user locations with one another for private line services, and to interconnect end users with the VPLS and IP/MPLS service edges for enterprise, residential, and mobility services. OAM and protection capabilities are critical to enable Ethernet to be deployed as an infrastructure technology. This article reviews several important developments in Ethernet OAM and protection standards, and discusses how those capabilities are vital for the creation of an effective Ethernet infrastructure. [ABSTRACT FROM AUTHOR]

In variable-speed electrical drives, data transmission between an electric motor and an inverter is required because of sensors installed at the motor. Sensor information is used both in the motor control and in the diagnostics. For example, a data transmission medium can be implemented either by additional cabling or through the power cable. However, additional cabling is neither a cost-effective nor even a reliable solution. Previously, only a few application-specific solutions for data transmission via power cables have been presented. A general method to implement a broadband and Ethernet-based communication medium between a motor and an inverter is proposed here. The method forms an Ethernet-supported packet-based communication medium over the motor power cable. Ethernet supports all kind of protocols implemented above the link layer. The channel and noise characteristics are described. According to these, a coupling interface is developed that allows installing the communication device safely to a three-phase inverter-fed motor power cable. A channel capacity analysis is also performed. Laboratory measurements are carried out and analysed with the proposed method, and two possible applications are discussed. [ABSTRACT FROM AUTHOR]

ISO-16484-5 (EN ISO 2003) and ANSI/ASHRAE Standard 135-2004 (BACnet®) is a standard data communication protocol for building automation and control systems. In this study, the performance of three BACnet local area networking options is investigated with simulation models developed using ARENA, a tool for simulating discrete event dynamic systems. This study evaluates the delay characteristics of Master-Slave/Token-Passing (MS/TP), ANSI/ATA-878. 1 (ARC- NET), and ISO-8802-3 (Ethernet) medium access control (MAC) layers being used to deliver BACnet application services. Analysis of the simulation results was used to develop recommendations that should be considered when designing and operating BACnet systems. [ABSTRACT FROM AUTHOR]

Enterprises are increasingly using Ethernet as the foundation for transforming their networks to Internet Protocol. Simultaneously, service providers are deploying Ethernet to exploit the demand for wide-area Ethernet services and as the infrastructure for new residential services. such as IPTV. This is due to Ethernet's low cost per bit and ubiquity in local area networks. Recent years have seen the widespread deployment of IP/MPLS networks to address this opportunity. IP/MPLS enables enhanced flexibility over the same converged network for IP and legacy services, avoiding the need to build separate per-service networks. It also adds carrier-grade capabilities such as quality of service, traffic engineering, and resiliency, thereby enabling new multipoint services such as virtual private LAN service. However, using Ethernet for "always-on" and other mission-critical services has resulted in new resiliency requirements, in both the access and the network core. Two novel developments address these high expectations by enabling significant improvements in service availability. These are pseudowire redundancy and Ethernet multi-chassis link aggregation. This article reviews the current redundancy mechanisms typically deployed in Ethernet and MPLS networks. We show how additional enhancements are required in both the network core and the access to the Ethernet service. We describe new pseudowire redundancy and MC-LAG mechanisms, showing how they work together to enable end-to-end protection for Ethernet virtual private wire services and VPLS. [ABSTRACT FROM AUTHOR]

In 1995, the IEEE approved the 802.12 standard for data transmission at 100-Mbit/s using the Demand Priority Net- work Access protocol. 100VG-AnyLAN products conforming to this standard offered an upgrade path for Ethernet and Token Ring networks, without requiring new building wiring. A key factor in the approval of the 802.12 standard was the demonstrated error detection properties of its coding scheme. In particular, the coding scheme allows the detection of error bursts affecting encoded data carried on four parallel conductors, using nothing more than the standard IEEE 32-bit cyclic redundancy check applied to the unencoded data. Although these error detection properties were presented for verification as part of the standards process, for many years commercial considerations prevented public disclosure of how the code was actually found. These considerations no longer apply, and, in this paper, we explain in detail the design principles of the code, combining geometrical insight, linear algebra, combinatorial reasoning, and computer search. [ABSTRACT FROM AUTHOR]

This paper investigates the issue of fairness in IEEE 802.11-based multirate wireless local area networks (WLANs). Distributed coordination function, which is the medium-access-control (MAC) protocol used in 802.11 WLANs, provides equal long-term channel access probability to competing stations, irrespective of the time required in sending a frame. When equal-sized frames are used and channel conditions are similar, each station, regardless of its data rate, achieves the same throughput. Furthermore, the aggregate throughput is reduced to a level much closer to what one gets when all stations are of lower rate. This anomaly in the performance is a result of unfair channel time allocation for stations when they use multiple data rates. We consider provisioning of time-based fairness in which each station receives an equal share of the wireless channel occupancy time. We demonstrate that under time-based fairness, the throughput that a tagged node achieves in a multirate WLAN with n nodes is identical to what it would achieve in a single-rate WLAN with n nodes all at the same data rate as the tagged node. Furthermore, we show that under time-based fairness scheme, the ratio of throughputs per station corresponding to two different bit rates is directly proportional to the ratio of their bit rates. We analyze different mechanisms in achieving time-based fairness by using an analytical model. Using Jam's fairness index, optimal MAC parameters required in achieving maximum fairness between slow and fast stations are obtained. The impacts of these mechanisms on throughput of slow and fast stations are explored. We also consider the notion of proportional fairness in a multirate scenario and prove that it is equivalent to fair channel time allocation. Last, our investigation of an alternative fairness criterion also leads us to propose that the IEEE 802.11 MAC protocol should be redesigned with temporal fairness as a design objective in avoiding inefficiencies related to the performance anomaly. [ABSTRACT FROM AUTHOR]

Abstract: Bandwidth demands on ubiquitous Ethernet have grown immensely, driven by the rapid expansion of real-time applications like audio/video streaming. In a related research, the authors designed a novel high-performance custom network processor chip using field programmable gate arrays (FPGAs). The main function of this chip (named SPEED) is to bypass the operating system processing of network protocol stack at the host computer by off-loading its networking functions to hardware. This simplification is not only required to fit in a hardware solution, but also it improves network performance. The novel chip utilizes the concept of Ethernet channels, where Ethernet frames are addressed in a multicast addressing scheme. In this article, we integrate the chip with a Firewire peripheral interface (FPI). A crucial function of the FPI design is to convert the IEEE 1394 isochronous traffic to the Ethernet frame format via two independent asynchronous write and read buffers. The FPI also manages the SPEED hardware interrupt signals in a bi-directional communication scheme. The goal of this research is to map isochronous Firewire packets into Ethernet frames by utilizing the SPEED Ethernet channel assignment capability. We used Verilog Hardware Description Language (HDL) to synthesize the FPI design. Since the FPI needs to support bi-directional communication, we also present a generic HDL model for bi-directional data sharing, which can be used in similar bi-directional applications. Initial performance measures show that the FPI consumes less that 0.15W of power. Also we found that the synthesized design utilizes only 28% of the target chip resources. Hence it was possible to incorporate it with the SPEED design in the same FPGA chip. This low power consumption will lead to powering SPEED-FPI based network devices through the network cable and eliminate the process of regular electrical power outlet installations and maintenance. Thus, the SPEED-FPI system reduces the installation complexity, especially for large number of devices (e.g., surveillance cameras). [Copyright &y& Elsevier]

The results of a performance analysis of mobile agents in a distributed network environment are presented. Analytical models are developed and used to compare the performance of carrying out a simple management function with mobile agents (MA) or SNMP (simple network management protocol). An examination is made of how delay is affected by the size of the management domain and how each method behaves in both a local area network (LAN) and wide area network (WAN) environment. For large domains, optimal performance is obtained if the mobile agents' size is bounded, and this may require sharing the management task among a small number of mobile agents. [ABSTRACT FROM AUTHOR]

The growth of multimedia applications on the Internet made at least one fifth of the total network traffic to run over UDP. Unlike TCP, UDP is unresponsive to network congestion. This may cause, inter alia, bandwidth starvation of responsive flows, severe and prolonged congestions or, in the worst-case scenario, a congestion collapse. Hence, the coexistence of both protocols on fair-share premises converges towards impossibility. The paper deals with a new approach to solving the problem of taming down the unresponsive flows. By using some of the desirable properties of mobile agents, the system is able to control the influx of non-TCP or unresponsive flows into the network. Various functions performed by mobile agents monitor non-TCP flows, calculate sending rates and modify their intensity according to the needs of the network to attain as good performance as it is possible. [ABSTRACT FROM AUTHOR]

We propose an extension of the classical CSMA/CD protocol that eliminates its three main drawbacks. The new protocol, called dual-mode CSMA/CD (CSMA/DM), operates in two modes, light mode and heavy mode. The light mode of CSMA/DM is almost the same as the original CSMA/CD protocol and is primarily used when the LAN load is light. The heavy mode is a collision free mode and is applied when the LAN load is heavy. The proposed modification to CSMA/CD is minimal while performance gain is significant. CSMA/DM automatically switches between its two modes based on the observed LAN load. Under heavy mode, the monitor station in a CSMA/DM LAN allocates bandwidth to those busy stations without collisions. The newly added priority scheme in the heavy mode allows stations to reserve bandwidth for their high priority frames. Compared with the current collision-free fast Ethernet and Gigabit Ethernet, CSMA/DM has the advantage of not needing any extra switches or hubs while maintaining comparable performance. Its priority scheme provides more flexibility on bandwidth distribution than in fast Ethernet and Gigabit Ethernet. It can also be easily adapted for high-speed wireless LANs. More importantly, CSMA/DM should be a good alternative of the widely used CSMA/CA (collision avoidance) in mobile ad hoc networks (MANET) and sensor networks. Copyright © 2005 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

We study a price-based rate control mechanism for random access networks. The mechanism uses channel feedback information to control the aggregate packet arrival rate. For our analysis, we use the standard slotted Aloha model with an infinite set of nodes. We show that the resulting Markov chain is positive recurrent. In addition, we characterize the throughput and delay at the operating point of the system and show how the operating point can be set a priori by appropriately choosing the control parameters. We illustrate our results using numerical experiments. [ABSTRACT FROM AUTHOR]

Focuses on the evolution of communication methods used in industrial application; Description of the benefits that CAN offers for industries; Advantages of using TCP/IP over Ethernet; Development of industrial-grade connectors and cabling for Ethernet.

A novel statistical model, the one-dimensional hyperbolic distribution, is proposed for network traffic activity. This includes the modelling of network traffic performance and descriptive parameters, organised as probability distribution functions of activity monitored during an observation window. This model promises accurate representation in a compact manner, of particular interest in statistical anomaly intrusion detection systems. The hyperbolic distribution has been tested in modelling several performance parameters in data traces of IP traffic collected from various actual network environments, including the campus LANs of the New Jersey institute of Technology and the WAN of the Bergen County Cooperative Library Network System (BCCLNS), as well as simulated client-server LANs. By fitting the model parameters to the experimental data, it is shown that this analytic model performs significantly better than the models currently in common use, such as the Weibull, Pareto, normal and lognormal distributions. [ABSTRACT FROM AUTHOR]

The boundaries between packet and circuit networks have long disappeared, with many traditional circuit-switched applications such as voice and video now being carried over packet-switched IP/MPLS or Ethernet networks. However, this transition has happened so fast that many OAM functions supported by circuit-switched networks such as SONET/SDH are still unmatched in packet networks. In order to match the quality sustained by circuit-switched networks, OAM functions must also be developed for such packet networks. A number of recent efforts have started to address OAM functions for IP/MPLS and Ethernet packet technologies. Service providers and carriers alike are the driving force behind the work, as there is general recognition that to generate sustained revenues, services must be efficiently managed. In this article we discuss issues in providing OAM features and capabilities for MPLS-based packet networks. [ABSTRACT FROM AUTHOR]

Recent research has shown that long-range dependence is found not only in network traffic volume, but also in the propagation delay time of packets. The round-trip propagation time of packets, which is known as the round-trip time (RTT), is one of the most important parameters for determining TCP performance. The ability to understand its characteristics can contribute to an improvement in TCP performance. Therefore, in this paper, to determine the impact of the long-range dependence of the RTT on TCP performance, the authors looked at the two congestion control phases known as congestion avoidance and slow start in TCP window control and used computer simulations to evaluate performance. First, the evaluation related to the impact on the congestion avoidance phase showed that higher long-range dependence and larger transfer files caused greater variability in the file transfer completion time. Next, the evaluation related to the impact on the slow start phase, which evaluated the impact of the long-range dependence of the RTT on the TCP retransmission timeout (RTO), showed that higher long-range dependence causes unnecessary timeouts, which end up causing the resending of packets that have not actually been lost, to occur more frequently. The authors also performed investigations for mitigating the impact of the long-range dependence of the RTT and made a number of modifications to the existing RTO estimation algorithm to show that TCP performance can be improved. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(6): 52–60, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10073 [ABSTRACT FROM AUTHOR]

Using the upgrading of the measuring system of the GOL-3 plasma facility as an example, the structure of hardware and software environment developed for data-acquisition systems of large pulse electrophysical plants is considered. The equipment is built into 3U-size 19-inch subracks. The control computers are connected to the measuring equipment via standard network facilities (Ethernet-10/100 interface and TCP/IP protocol). At present, we have begun making basic diagnostic measurements using the ADC1250/32 multichannel synchronous data-acquisition systems. [ABSTRACT FROM AUTHOR]

The shared media local area networks (LAN) such as Ethernet/FDDI has the bottleneck of bandwidth scalability. ATMLAN based on asynchronous transfer mode (ATM) switch is considered interesting at present as the next-generation multimedia LAN to dissolve this bottleneck. The research and development of ATM has mostly been promoted for application to the public network. With respect to LAN, a new investigation is needed which includes the architecture since there are various requirements such as compatibility to the existing protocol, the simplification of the network control, and the drastic reduction of the system cost. This paper proposes the ATM-oriented multimedia LAN (ATOMLAN) architecture which meets the forementioned requirements. The proposed architecture has the following three features: (I) the ATMLAN addressing, considering the affinity to the existing protocol and the efficient internetworking; (2) the high-speed connection control to realize a high throughput for the connectionless-type data transfer; and (3) the simple and efficient traffic control for the high-speed burst-multimedia service. Using the proposed architecture, a seamless high-speed multimedia network can be realized, from the desk-top to the broadband network. [ABSTRACT FROM AUTHOR]

Computing systems are evolving into very large global networks that interconnect competing individuals, organizations, and even countries. We present a secure network protocol called Authenticated Datagram Protocol (ADP) that optimizes performance by establishing host-to-host secure channels and building agent-to-agent channels on top of host-to-host channels. However, the performance advantages of ADP come with an accompanying set of trust requirements that are noticeably stringent for a network spanning mutually distrustful organizations. We identify the cause for this stringency to be propagation of trust relationships in ADP and present methods to break their propagation, thereby accomplishing a significant reduction in ADP's trust requirements. ADP, being a protocol for establishing host- to-host channels, can be handled at the subtransport level of the protocol hierarchy. A prototype of ADP has been implemented on Sun workstations connected by an Ethernet. Experimental measurements confirm that both the average latency of messages and the maximum throughput are substantially better than other secure protocols. [ABSTRACT FROM AUTHOR]

Communication protocols provide facilities that may be used by processes to intercommunicate across machine boundaries. In this context, performance evaluations serve as guidelines to provide efficient behavior for distributed applications as characterized by critical transfer time constraints. Since data transfers in actual operating environments are carried out on preexistent intermachine communications, the effect of loaded communication media on performance is of interest. This paper presents a study on the performance achieved in the presence of variable background load when applications are directly based on the transport level. The analysis was carried out through experiments over an Ethernet local area network and was repeated on two different protocol suites, TCP/IP and XNS. Communications by means of datagram and reliable transfer primitives were analyzed for both protocol structures. The dependency of protocol performance on background load, as well as the localization of the most important causes affecting throughput as load increases, were assessed. Different strategies to reduce the impact of background load are proposed on the basis of the results acquired so far. [ABSTRACT FROM AUTHOR]

We propose an interface for use from within UNIX' user programs for communicating over multiple and varied local and wide area networks. This interface aids the design of a distributed application program by hiding the actual communications protocols used over each network, and providing instead simple primitives for sending and receiving (possibly large) datagrams, using a simple standardized network addressing scheme based on a pair. This interface has been designed principally for use by the Newcastle Connection, a software subsystem which is used to construct the UNIX United distributed system; however, this interface may well fulfill the needs of a large class of transaction-oriented applications of UNIX. Comparisons are made with various alternative approaches to network interfacing, including the Xerox Pup communications architecture, the "Clean and Simple" approach of University College London, and the socket scheme developed for Berkeley UNIX. [ABSTRACT FROM AUTHOR]

This paper presents a new protocol for token ring local computer networks and its performance for single and dual ring networks using simulation modelling. The proposed protocol is a modification of the IEEE 802.5 token ring standard. In this protocol, a station can start transmission if it receives a free token or a data packet passing its interface logic. Idle stations are skipped whenever there is at least one station that has a ready packet to transmit. This is achieved by using two of the reserved bits in the frame status (PS) field. The bits used are called transmission reservation bits (TR-bits). The TR-bits are reset by the sending station and set by the first station that has data to transmit during the round trip of the associated packet along the ring. It is found that the proposed modified token ring (MTR) protocol provides better throughput and delay performance for both the single and dual ring networks. The reduction in mean delay of the MTR as compared to the standard token ring (STR) reaches about 45 per cent under heavy traffic conditions. Moreover, it is found that the performance of the MTR decreases as the packet size increases but remains higher than the corresponding performance of the STR for both the single and dual ring networks. The MTR provides a maximum throughput improvement of about 6 per cent over the STR for both dual and single ring cases. Finally, the proposed protocol (MTR) is easy to implement, with no extra hardware and has low cost! performance characteristics. [ABSTRACT FROM AUTHOR]

CSMA protocols have been extensively studied under the assumption of an equilibrium condition. The results obtained in this manner provided useful insight for the development of packet radio and local area networks for the transmission of bursty data. The integration of isochronous and bursty traffic in the same network may, however, make the equilibrium assumption unrealistic, for example when fixed-length frames are defined, only a portion of which is devoted to the transmission of bursty data according to a CSMA protocol. In such cases, it is necessary to study the protocol behaviours in fixed-length time segments. This paper presents a technique for the analysis of slotted CSMA protocols in such an environment. Definitions for the transient throughput and transient average packet delay are introduced, and closed form expressions are derived for non-persistent and 1-persistent CSMA and CSMA/CD protocols with arbitrary distribution of the packet length. Numerical results help in the visualization of some interesting phenomena, peculiar to the CSMA operations in finite-length intervals. [ABSTRACT FROM AUTHOR]

Local am network (LAN) is used widely at pres- eat, and the radio LAN also is utilized worldwide in practice. This paper focuses on the performance evaluation of the radio LAN in actual operation. It is shown that the method based (Al simple network management protocol (SNMP) is useful. [ABSTRACT FROM AUTHOR]

This paper evaluates the performance of a fast select connection oriented (FS-CO) logical link control (LLC) protocol in slotted ring networks. The FS-CO-LLC protocol is one of light-weight protocols. Is employs a method of implicit connection establishment and release to reduce overhead of connection control. This paper analyzes the performance (throughput and average response time) of a slotted ring network in which a single slot circulates by means of Equilibrium Point Analysis. The performance in the case of plural slots is evaluated by simulation. This paper also compares the FS-CO-LLC protocol with connection-oriented (CO) and connection-less (CL) LLC protocols. Numerical results show that the performance of FS-CO-LLC and CL-LLC protocols is superior to that of CO-LLC protocol when the average message length is smaller than the buffer size. On the other hand, when the average message length is larger than the buffer size, CO-LLC protocol has the best performance. It is also shown that the performance of FS-CO-LLC protocol does not degrade but CL-LLC protocol does, even if the average message length becomes larger. [ABSTRACT FROM AUTHOR]

This paper considers the logical link control (LLC) sublayer in the CSNAICD local area network, and compares analytically the performance of connection-oriented (CO) and connectionless (CL) protocols. As the medium access protocol, slotted nonpersistent CSMA/CD with a priority assigned to some types of frames is employed. In comparison, the overhead for connection establishment/ release in the CO-LLC protocol, and the receiver-buffer overflow in the CL-LLC protocol are especially considered, and the two kinds of protocols are compared. In the CO-LLC protocol, a connection is established corresponding to a message. Employing the technique of equilibrium point analysis, the throughput and the average response time of the two kinds of MAC protocols are derived. To compare the performance of the two kinds of protocols, the receiver-buffer size is kept constant, and the average number of data frames in a message is varied. As a result, it is shown that the CL-LLC protocol is better than the CO-LLC protocol when the average number of data frames is sufficiently less than the receiver-buffer size, but the order is reversed when the average number of data frames is increased. [ABSTRACT FROM AUTHOR]

This paper describes the algebraic specification with a high degree of abstraction for the token ring LAN protocol provided in IEEE standard 802.5. Then the anticipated properties are verified by an algebraic method, the specification satisfies the safeness property (any frame transmitted from the active station either has been received by all stations or received by consecutive stations in the downstream without an error) and the mutual exclusion property (there exists one and only one token on the ring). The verification is carried out interactively using the verification support system. The process of selecting the lemmas to be introduced, the process of verification, CPU time used for the verification and the number of days needed in the verification are discussed. [ABSTRACT FROM AUTHOR]

As the bandwidth-delay product of ring networks increases, conventional access protocols. such as the gated token protocol. become less efficient. We propose overcoming this problem with contention-based access protocols. We discuss the necessary properties of any contention ring access protocol and present two specific proposals. Some simple analysis and simulation is done to determine the performance of these protocols. Our protocols are shown to have better end-to-end delay than a gated token protocol. We also simulate another contention protocol called the hybrid protocol. proposed by Bhargava, et al. It is shown that the simple protocols described in this paper have performance comparable to that of the more complex hybrid protocol on networks with high bandwidth-delay products. [ABSTRACT FROM AUTHOR]

We introduce an improved Ethernet ring protection method, selective filtering database (FDB) advertisement, to minimize traffic overshoot in the event of a failure or recovery. On the protection switching event, the proposed method makes all ring nodes perform an FDB flush except the FDB entries associated with their client subnets. Then, ring nodes rapidly exchange their client MAC address lists so that their FDBs are immediately updated by indirect MAC address learning. The proposed scheme guarantees fast and reliable protection switching over the standard scheme. [ABSTRACT FROM AUTHOR]

Simulation of ETSI's High Performance Local Area Network Type 2 (HIPERLAN/2) is presented. MATLAB/Simulink modelling demonstrated that for 64-QAM mode, with coding rate 2/3, a data rate of 48 Mbit/s is achieved with 3.2 dB SNR improvement compared to coding rate 3/4 (with puncturing) as required for the maximum 54 Mbit/s. [ABSTRACT FROM AUTHOR]

Reports on the growth of wireless local area network connectivity in the U.S. Growth of wireless data market; Plan of McDonald's to launch its own Wi-Fi service.

Anritsu Co., unveiled at Supercomm the MP159OB network performance tester, the first single test instrument capable of conducting Ethernet over SONET/SDH (EoS), Ethernet/IP, Virtual Concatenation, 0Th, and SONET/SDH tests. Designed to meet the needs of next generation optical networks and network elements such as switches and routers, the MP159OB provides unique simultaneous analysis on multiple interfaces up to 10.7G. This means, for example, that the MPI59OB can perform full 0Th, SONET, or EoS analysis on one interface while analyzing the same or different data on one or more Ethernet/IP interfaces. The MP159OB can perform all major EoS tests, including Generic Framing Procedure, Local Area Network Extension Protocol for PPP, and Link Access Procedure. In an EoS application, for example, the network performance tester can check encapsulation functions and throughput via simultaneous operation on Ethernet and SONET/GFP links while checking errors/alarms of both links utilizing 15 GFP counters, 40+ SONET/SDFI counters, and 60+ Ethernet/IP counters. MPI59OB can test virtual concatenation and the Link Capacity Adjustment Scheme protocol as well.

The article focuses on some proposals for the use of real-time Ethernet in industries. Ethernet is starting to replace the industrial fieldbuses that were developed in the 1990s. There are now three proposals for real-time Ethernet. Two are from German companies: Siemens (PROFInet v3) and Beckhoff (EtherCAT ) and the third is from B&R Automation (Powerlink), based near Salzburg. All three rely on the basic Ethernet frame or "telegram" for communication. No high level protocol like TCP/IP is used, although Ethernet and TCP/IP may co-exist on the same cable.